In a communication system a communication network is provided, which can link together two communication terminals so that the terminals can send information to each other in a call or other communication event. Information may include speech, text, images or video.
Modern communication systems are based on the transmission of digital signals Analogue information such as speech is input into an analogue to digital converter at the transmitter of one terminal and converted into a digital signal. The digital signal is then encoded and placed in data packets for transmission over a channel to the receiver of another terminal.
Each data packet includes a header portion and a payload portion. The header portion of the data packet contains data for transmitting and processing the data packet. This information may include an identification number and source address that uniquely identifies the packet, a header checksum used to detect processing errors and the destination address. The payload portion of the data packet includes information from the digital signal intended for transmission. This information may be included in the payload as encoded frames such as video frames, wherein each frame represents a portion of the video signal.
One type of communication network suitable for transmitting data packets is the internet. Protocols which are used to carry voice signals over an Internet Protocol network are commonly referred to as Voice over IP (VoIP). VoIP is the routing of voice conversations over the Internet or through any other IP-based network.
Conditions associated with the communication system, such as resource availability of communication terminals can affect the ability of the terminals to process data. For example, CPU (central processing unit) resources will affect how effectively the transmitting and receiving terminals can process information.
It is therefore necessary to optimise the manner in which information is processed and transmitted by the terminals in accordance with the conditions associated with the communication system.
In a known solution a receiving terminal may report to the transmitting terminal the CPU resource of the receiving terminal that is available to process the information received from the transmitting terminal. The transmitting terminal may then adjust the rate at which data is transmitted to the receiving terminal in dependence on the available CPU resource of the receiving terminal.
However during a two way communication event such as a video call, where each terminal transmits and receives video data, the users of each terminal may find that they experience a different quality of service during the call. For example, whilst one terminal receives a high quality signal the other terminal may receive a low quality signal. This is disadvantageous for both participants of the call since the call is likely to be terminated if even one participant of the call experiences poor call quality.
It is also known to include an “Offer-Answer” model in an internet protocol, whereby a receiving terminal can send back a request to a transmitting terminal requesting a certain bitrate signal. This allows the receiving terminal to negotiate the streaming it receives. However, the current standard is centred only around one-way transmission and does not specify how the requested bitrate is to be determined. Hence if used during a two-way communication such as a voice or video call then the current model will in fact treat the two signals of the call (from first to second terminal and vice-versa) as two completely separate one-way communications, i.e. so the two streams of the call will be negotiated separately.
It is an aim of the present invention to achieve a balanced quality of service among the participants of a communication event and to overcome the above identified problems.